Sorry, The way I recommend doing this was assuming the user on the Asterisk box needed to be publicly reachable from anywhere.
I think that approach makes sense when using DID's and inbound routing that does need authentication. On 5/4/10 12:55 PM, Olle E. Johansson wrote: > 4 maj 2010 kl. 18.30 skrev Brett Nemeroff: > > >> Carmelo, >> If you have an SIP peer that matches the host and port of the opensips >> server.. ie: >> [opensips] >> type=friend >> host=<ip of opensips. >> port=<port of opensips> (can be omitted if port 5060) >> >> Then it'll match that.. typically if it's coming from opensips you'll want >> to add: >> insecure=invite >> >> so that opensips won't be challenged to authenticate. Also be sure there is >> no secret set. >> >> I personally wouldn't do this using the default context as the other posters >> had recommended as that will allow *anyone* to send traffic to your asterisk >> server. Which I don't believe is what you really want to do. Instead, create >> a peer that is limited by IP and PORT allowed to send invites without a >> secret. >> >> Also be sure that the context for that peer is set to the right context and >> that if from the asterisk CLI you type: >> dialplan show<RURI username>@<opensips context> >> that it matches something you'd expect. >> >> On another note, are you performing a consume credentials? I think it >> *might* be possible that opensips is forwarding your UAC's credentials on to >> Asterisk if you are not.. >> >> > If you want to ONLY match on IP/port, you need to use "type=peer". > > regards, > /O > > >> -Brett >> >> >> On Tue, May 4, 2010 at 8:02 AM, wüber<leon...@gmail.com> wrote: >> >> Hi Bogdan, >> >> connecting Opensips with Asterisk I can see that if a client registered on >> Opensips server tries to make a call to a client in Asterisk domain, after >> the INVITE, it receives a "forbidden" message from asterisk. I have set the >> forwarding functionality in Opensips (rewriteuri function) and I'm pretty >> sure it's something related to asterisk. >> >> Perhaps this is not the right section, but anyway could you help me? Do you >> know what I should set in the sip.conf of Asterisk config file? >> >> Thanks a lot, >> Carmelo >> -- >> View this message in context: >> http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-and-asterisk-tp4962200p5003181.html >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > --- > * Olle E Johansson - o...@edvina.net > * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden > > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
